Acoustics Speech and Signal Processing 1988 Icassp 88 1988 International Conference on, Mar 1, 2008
Artificial bandwidth extension techniques can be employed in mobile terminals to improve the qual... more Artificial bandwidth extension techniques can be employed in mobile terminals to improve the quality of the far-end speaker's signal at the receiver. To accomplish this, usually statistical models are trained requiring wideband speech material from a language that is expected to be used in the conversation. In practice however, the language of a certain phone conversation is not known to the user equipment. Therefore we investigated the performance of an HMM- based multilingually trained artificial bandwidth extension on speech signals of which the language was unseen in training. The cross-language training and test turned out to cause only minor degradations compared to the use of monolingually trained acoustic models of the language used in test. Our findings indicate that artificial bandwidth extension can be efficiently trained with multilingual speech data without significant losses in speech quality.
Automatic speech recognition in mobile devices has to cope with varying acoustical background noi... more Automatic speech recognition in mobile devices has to cope with varying acoustical background noises in potentially low SNR situations. Its performance in car noise environments is of our particular interest. We put focus on noise reduction techniques as ...
2000 IEEE Workshop on Speech Coding. Proceedings. Meeting the Challenges of the New Millennium (Cat. No.00EX421), 2000
In mobile speech communication usually vector quantization (VQ) is employed to ensure high coding... more In mobile speech communication usually vector quantization (VQ) is employed to ensure high coding efficiency. Convolutional coding and softbit speech decoding can add a considerable amount of robustness. VQ as well as softbit decoding however can be a quite complex task. Under the constraint of a constant gross bit rate and clean channel quality we propose the use of lower
Vehicular Technology Conference Fall 2000. IEEE VTS Fall VTC2000. 52nd Vehicular Technology Conference (Cat. No.00CH37152), 2000
Speech, audio, image or video signals are highly correlated and source coding algorithms are used... more Speech, audio, image or video signals are highly correlated and source coding algorithms are used to reduce the redundancy. Due to complexity and delay constraints the coded signal contains redundancy both inside one frame and also from frame to frame (time correlation). There has been increasing interest in algorithms that exploit the a priori knowledge within the channel or the source decoder. We present a unified view of source and channel decoding supported by a priori knowledge. The complexity of our decoding algorithm depends exponentially on the number of bits representing a quantized symbol. Therefore we investigate source encoding by scalar quantization. With this concept more bits per dimension are required for source encoding because we have to compensate for the loss in coding gain introduced by the lower dimensionality. As a consequence, less redundancy can be added by a channel code if the gross bit rate is kept constant. However the lower dimensionality allows the efficient use of unequal error protection (UEP) and our proposed decoding techniques. We show the application of our joint source-channel (de-)coding concepts to the ANSI-136 speech codec
Speech coding at low and medium bit rates benefits from the assumption that the speech signal is ... more Speech coding at low and medium bit rates benefits from the assumption that the speech signal is stationary inside a subframe or even inside consecutive subframes. Speech signals however comprise a considerable amount of transient segments, such as onsets, which are usually difficult to encode with good quality. This paper proposes a method to improve the encoding of transient (sub-)frames by disregarding the adaptive codebook contribution and strengthening the fixed codebook contribution. This can principally be applied to any CELP-like standard speech coder while keeping the same bit rate. The proposed technique has been successfully employed in the ITU-T 4 kbps speech coder candidate commonly proposed by AT&T, Conexant currently Mindspeed, Deutsche Telekom, France Telecom, Matsushita, NTT, and Siemens in November 2001. Simulation results show the improvements achieved by this approach.
Acoustics Speech and Signal Processing 1988 Icassp 88 1988 International Conference on, Mar 1, 2008
Artificial bandwidth extension techniques can be employed in mobile terminals to improve the qual... more Artificial bandwidth extension techniques can be employed in mobile terminals to improve the quality of the far-end speaker's signal at the receiver. To accomplish this, usually statistical models are trained requiring wideband speech material from a language that is expected to be used in the conversation. In practice however, the language of a certain phone conversation is not known to the user equipment. Therefore we investigated the performance of an HMM- based multilingually trained artificial bandwidth extension on speech signals of which the language was unseen in training. The cross-language training and test turned out to cause only minor degradations compared to the use of monolingually trained acoustic models of the language used in test. Our findings indicate that artificial bandwidth extension can be efficiently trained with multilingual speech data without significant losses in speech quality.
Automatic speech recognition in mobile devices has to cope with varying acoustical background noi... more Automatic speech recognition in mobile devices has to cope with varying acoustical background noises in potentially low SNR situations. Its performance in car noise environments is of our particular interest. We put focus on noise reduction techniques as ...
2000 IEEE Workshop on Speech Coding. Proceedings. Meeting the Challenges of the New Millennium (Cat. No.00EX421), 2000
In mobile speech communication usually vector quantization (VQ) is employed to ensure high coding... more In mobile speech communication usually vector quantization (VQ) is employed to ensure high coding efficiency. Convolutional coding and softbit speech decoding can add a considerable amount of robustness. VQ as well as softbit decoding however can be a quite complex task. Under the constraint of a constant gross bit rate and clean channel quality we propose the use of lower
Vehicular Technology Conference Fall 2000. IEEE VTS Fall VTC2000. 52nd Vehicular Technology Conference (Cat. No.00CH37152), 2000
Speech, audio, image or video signals are highly correlated and source coding algorithms are used... more Speech, audio, image or video signals are highly correlated and source coding algorithms are used to reduce the redundancy. Due to complexity and delay constraints the coded signal contains redundancy both inside one frame and also from frame to frame (time correlation). There has been increasing interest in algorithms that exploit the a priori knowledge within the channel or the source decoder. We present a unified view of source and channel decoding supported by a priori knowledge. The complexity of our decoding algorithm depends exponentially on the number of bits representing a quantized symbol. Therefore we investigate source encoding by scalar quantization. With this concept more bits per dimension are required for source encoding because we have to compensate for the loss in coding gain introduced by the lower dimensionality. As a consequence, less redundancy can be added by a channel code if the gross bit rate is kept constant. However the lower dimensionality allows the efficient use of unequal error protection (UEP) and our proposed decoding techniques. We show the application of our joint source-channel (de-)coding concepts to the ANSI-136 speech codec
Speech coding at low and medium bit rates benefits from the assumption that the speech signal is ... more Speech coding at low and medium bit rates benefits from the assumption that the speech signal is stationary inside a subframe or even inside consecutive subframes. Speech signals however comprise a considerable amount of transient segments, such as onsets, which are usually difficult to encode with good quality. This paper proposes a method to improve the encoding of transient (sub-)frames by disregarding the adaptive codebook contribution and strengthening the fixed codebook contribution. This can principally be applied to any CELP-like standard speech coder while keeping the same bit rate. The proposed technique has been successfully employed in the ITU-T 4 kbps speech coder candidate commonly proposed by AT&T, Conexant currently Mindspeed, Deutsche Telekom, France Telecom, Matsushita, NTT, and Siemens in November 2001. Simulation results show the improvements achieved by this approach.
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